<div dir="ltr">Yes, well, really is Elastix. Hmmm where I need to pt directmedia=no ?<div><br></div><div>Thanks,</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, Mar 10, 2014 at 2:38 PM, Eric Wieling <span dir="ltr"><<a href="mailto:EWieling@nyigc.com" target="_blank">EWieling@nyigc.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Try ulaw instead of g729, set directmedia=no<br>
<br>
I see you are using FreePBX. I cannot help further.<br>
<div class="im HOEnZb"><br>
<br>
-----Original Message-----<br>
From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of <a href="mailto:alpocr@gmail.com">alpocr@gmail.com</a><br>
</div><div class="HOEnZb"><div class="h5">Sent: Monday, March 10, 2014 4:15 PM<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
Cc: <a href="mailto:andres@telesip.net">andres@telesip.net</a><br>
Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds.<br>
<br>
Guys, hi. I have not solved the problem. Outgoing calls to remote extensions drops on 5-20 seconds. Incoming calls work perfectly.<br>
<br>
Hope you can help me. Attach updated logs: <a href="http://pastebin.com/HmKEDAUq" target="_blank">http://pastebin.com/HmKEDAUq</a><br>
<br>
Thanks,<br>
<br>
<br>
On Thu, Dec 19, 2013 at 8:48 AM, Eric Wieling <<a href="mailto:EWieling@nyigc.com">EWieling@nyigc.com</a>> wrote:<br>
<br>
<br>
See sip.conf.sample in the Asterisk tarball for documentation of valid settings.<br>
<br>
<br>
-----Original Message-----<br>
From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of <a href="mailto:alpocr@gmail.com">alpocr@gmail.com</a><br>
<br>
Sent: Wednesday, December 18, 2013 9:30 PM<br>
To: <a href="mailto:andres@telesip.net">andres@telesip.net</a>; Asterisk Users Mailing List - Non-Commercial Discussion<br>
Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds.<br>
<br>
<br>
I set canreinvite=very in the remote extension, and now the call not drops. Valid solution?<br>
<br>
<br>
On Wed, Dec 18, 2013 at 6:38 PM, Andres <<a href="mailto:andres@telesip.net">andres@telesip.net</a>> wrote:<br>
<br>
<br>
On 12/18/13, 3:09 PM, <a href="mailto:alpocr@gmail.com">alpocr@gmail.com</a> wrote:<br>
<br>
<br>
Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds.<br>
<br>
I got my my NAT firewall properly configured. Here I attached my debug in CLI: <a href="http://pastebin.com/gh34E69f" target="_blank">http://pastebin.com/gh34E69f</a><br>
<br>
<br>
When the call is setup I see your Asterisk retransmitting the "SIP/2.0 200 OK" packet many times and getting no response. The other end needs to receive the packet and generate an "ACK". You need to trace where that packet is going and figure out why it is not reaching its target, or if it is, then why is the ACK not making it back. Thats your problem.<br>
<br>
<br>
Thank you!<br>
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</div></div></blockquote></div><br><br clear="all"><div><br></div>-- <br><div dir="ltr">Allan Porras <div><a href="http://www.AllanPorras.com" target="_blank">http://allanPorras.com</a></div><div>Google Plus: <a href="http://goo.gl/BRkbX" target="_blank">http://goo.gl/BRkbX</a> <br>
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